diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc
index ac02482427..704ba7963e 100644
--- a/audio/channel_receive.cc
+++ b/audio/channel_receive.cc
@@ -437,7 +437,9 @@ void ChannelReceive::OnRtpPacket(const RtpPacketReceived& packet) {
     if (has_audio_level)
       last_received_rtp_audio_level_ = audio_level;
     std::vector<uint32_t> csrcs = packet.Csrcs();
-    contributing_sources_.Update(now_ms, csrcs);
+    contributing_sources_.Update(
+        now_ms, csrcs,
+        has_audio_level ? absl::optional<uint8_t>(audio_level) : absl::nullopt);
   }
 
   // Store playout timestamp for the received RTP packet
diff --git a/modules/rtp_rtcp/source/contributing_sources.cc b/modules/rtp_rtcp/source/contributing_sources.cc
index 853706c4f5..64dc443e29 100644
--- a/modules/rtp_rtcp/source/contributing_sources.cc
+++ b/modules/rtp_rtcp/source/contributing_sources.cc
@@ -25,9 +25,11 @@ ContributingSources::ContributingSources() = default;
 ContributingSources::~ContributingSources() = default;
 
 void ContributingSources::Update(int64_t now_ms,
-                                 rtc::ArrayView<const uint32_t> csrcs) {
+                                 rtc::ArrayView<const uint32_t> csrcs,
+                                 absl::optional<uint8_t> audio_level) {
+  Entry entry = { now_ms, audio_level };
   for (uint32_t csrc : csrcs) {
-    last_seen_ms_[csrc] = now_ms;
+    active_csrcs_[csrc] = entry;
   }
   if (!next_pruning_ms_) {
     next_pruning_ms_ = now_ms + kPruningIntervalMs;
@@ -43,9 +45,16 @@ void ContributingSources::Update(int64_t now_ms,
 // non-const.
 std::vector<RtpSource> ContributingSources::GetSources(int64_t now_ms) const {
   std::vector<RtpSource> sources;
-  for (auto& record : last_seen_ms_) {
-    if (record.second >= now_ms - kHistoryMs) {
-      sources.emplace_back(record.second, record.first, RtpSourceType::CSRC);
+  for (auto& record : active_csrcs_) {
+    if (record.second.last_seen_ms >= now_ms - kHistoryMs) {
+      if (record.second.audio_level.has_value()) {
+        sources.emplace_back(record.second.last_seen_ms, record.first,
+                             RtpSourceType::CSRC,
+                             *record.second.audio_level);
+      } else {
+        sources.emplace_back(record.second.last_seen_ms, record.first,
+                             RtpSourceType::CSRC);
+      }
     }
   }
 
@@ -54,15 +63,20 @@ std::vector<RtpSource> ContributingSources::GetSources(int64_t now_ms) const {
 
 // Delete stale entries.
 void ContributingSources::DeleteOldEntries(int64_t now_ms) {
-  for (auto it = last_seen_ms_.begin(); it != last_seen_ms_.end();) {
-    if (it->second >= now_ms - kHistoryMs) {
+  for (auto it = active_csrcs_.begin(); it != active_csrcs_.end();) {
+    if (it->second.last_seen_ms >= now_ms - kHistoryMs) {
       // Still relevant.
       ++it;
     } else {
-      it = last_seen_ms_.erase(it);
+      it = active_csrcs_.erase(it);
     }
   }
   next_pruning_ms_ = now_ms + kPruningIntervalMs;
 }
 
+ContributingSources::Entry::Entry() = default;
+ContributingSources::Entry::Entry(int64_t timestamp_ms,
+                                  absl::optional<uint8_t> audio_level_arg)
+    : last_seen_ms(timestamp_ms), audio_level(audio_level_arg) {}
+
 }  // namespace webrtc
diff --git a/modules/rtp_rtcp/source/contributing_sources.h b/modules/rtp_rtcp/source/contributing_sources.h
index b6201ce794..5e34539ce4 100644
--- a/modules/rtp_rtcp/source/contributing_sources.h
+++ b/modules/rtp_rtcp/source/contributing_sources.h
@@ -32,18 +32,25 @@ class ContributingSources {
   ContributingSources();
   ~ContributingSources();
 
-  // TODO(bugs.webrtc.org/3333): Needs to be extended with audio-level, to
-  // support RFC6465.
-  void Update(int64_t now_ms, rtc::ArrayView<const uint32_t> csrcs);
+  void Update(int64_t now_ms, rtc::ArrayView<const uint32_t> csrcs,
+              absl::optional<uint8_t> audio_level);
 
   // Returns contributing sources seen the last 10 s.
   std::vector<RtpSource> GetSources(int64_t now_ms) const;
 
  private:
+  struct Entry {
+    Entry();
+    Entry(int64_t timestamp_ms, absl::optional<uint8_t> audio_level);
+
+    int64_t last_seen_ms;
+    absl::optional<uint8_t> audio_level;
+  };
+
   void DeleteOldEntries(int64_t now_ms);
 
   // Indexed by csrc.
-  std::map<uint32_t, int64_t> last_seen_ms_;
+  std::map<uint32_t, Entry> active_csrcs_;
   absl::optional<int64_t> next_pruning_ms_;
 };
 
diff --git a/modules/rtp_rtcp/source/contributing_sources_unittest.cc b/modules/rtp_rtcp/source/contributing_sources_unittest.cc
index 8b22d26c94..5f1d8d3d11 100644
--- a/modules/rtp_rtcp/source/contributing_sources_unittest.cc
+++ b/modules/rtp_rtcp/source/contributing_sources_unittest.cc
@@ -30,7 +30,7 @@ TEST(ContributingSourcesTest, RecordSources) {
   ContributingSources csrcs;
   constexpr uint32_t kCsrcs[] = {kCsrc1, kCsrc2};
   constexpr int64_t kTime1 = 10;
-  csrcs.Update(kTime1, kCsrcs);
+  csrcs.Update(kTime1, kCsrcs, absl::nullopt);
   EXPECT_THAT(
       csrcs.GetSources(kTime1),
       UnorderedElementsAre(RtpSource(kTime1, kCsrc1, RtpSourceType::CSRC),
@@ -45,12 +45,12 @@ TEST(ContributingSourcesTest, UpdateSources) {
   constexpr uint32_t kCsrcs2[] = {kCsrc2, kCsrc3};
   constexpr int64_t kTime1 = 10;
   constexpr int64_t kTime2 = kTime1 + 5 * rtc::kNumMillisecsPerSec;
-  csrcs.Update(kTime1, kCsrcs1);
+  csrcs.Update(kTime1, kCsrcs1, absl::nullopt);
   EXPECT_THAT(
       csrcs.GetSources(kTime1),
       UnorderedElementsAre(RtpSource(kTime1, kCsrc1, RtpSourceType::CSRC),
                            RtpSource(kTime1, kCsrc2, RtpSourceType::CSRC)));
-  csrcs.Update(kTime2, kCsrcs2);
+  csrcs.Update(kTime2, kCsrcs2, absl::nullopt);
   EXPECT_THAT(
       csrcs.GetSources(kTime2),
       UnorderedElementsAre(RtpSource(kTime1, kCsrc1, RtpSourceType::CSRC),
@@ -65,12 +65,12 @@ TEST(ContributingSourcesTest, ReturnRecentOnly) {
   constexpr int64_t kTime1 = 10;
   constexpr int64_t kTime2 = kTime1 + 5 * rtc::kNumMillisecsPerSec;
   constexpr int64_t kTime3 = kTime1 + 12 * rtc::kNumMillisecsPerSec;
-  csrcs.Update(kTime1, kCsrcs1);
+  csrcs.Update(kTime1, kCsrcs1, absl::nullopt);
   EXPECT_THAT(
       csrcs.GetSources(kTime1),
       UnorderedElementsAre(RtpSource(kTime1, kCsrc1, RtpSourceType::CSRC),
                            RtpSource(kTime1, kCsrc2, RtpSourceType::CSRC)));
-  csrcs.Update(kTime2, kCsrcs2);
+  csrcs.Update(kTime2, kCsrcs2, absl::nullopt);
   EXPECT_THAT(
       csrcs.GetSources(kTime3),
       UnorderedElementsAre(RtpSource(kTime2, kCsrc2, RtpSourceType::CSRC),
@@ -84,18 +84,18 @@ TEST(ContributingSourcesTest, PurgeOldSources) {
   constexpr int64_t kTime1 = 10;
   constexpr int64_t kTime2 = kTime1 + 10 * rtc::kNumMillisecsPerSec;
   constexpr int64_t kTime3 = kTime1 + 20 * rtc::kNumMillisecsPerSec;
-  csrcs.Update(kTime1, kCsrcs1);
+  csrcs.Update(kTime1, kCsrcs1, absl::nullopt);
   EXPECT_THAT(
       csrcs.GetSources(kTime2),
       UnorderedElementsAre(RtpSource(kTime1, kCsrc1, RtpSourceType::CSRC),
                            RtpSource(kTime1, kCsrc2, RtpSourceType::CSRC)));
-  csrcs.Update(kTime2, kCsrcs2);
+  csrcs.Update(kTime2, kCsrcs2, absl::nullopt);
   EXPECT_THAT(
       csrcs.GetSources(kTime2),
       UnorderedElementsAre(RtpSource(kTime1, kCsrc1, RtpSourceType::CSRC),
                            RtpSource(kTime2, kCsrc2, RtpSourceType::CSRC),
                            RtpSource(kTime2, kCsrc3, RtpSourceType::CSRC)));
-  csrcs.Update(kTime3, kCsrcs2);
+  csrcs.Update(kTime3, kCsrcs2, absl::nullopt);
   EXPECT_THAT(
       csrcs.GetSources(kTime3),
       UnorderedElementsAre(RtpSource(kTime3, kCsrc2, RtpSourceType::CSRC),
@@ -108,4 +108,22 @@ TEST(ContributingSourcesTest, PurgeOldSources) {
                            RtpSource(kTime3, kCsrc3, RtpSourceType::CSRC)));
 }
 
+TEST(ContributingSourcesTest, AudioLevel) {
+  ContributingSources csrcs;
+  constexpr uint32_t kCsrcs[] = {kCsrc1, kCsrc2};
+  constexpr int64_t kTime1 = 10;
+  csrcs.Update(kTime1, kCsrcs, 47);
+  EXPECT_THAT(
+      csrcs.GetSources(kTime1),
+      UnorderedElementsAre(RtpSource(kTime1, kCsrc1, RtpSourceType::CSRC, 47),
+                           RtpSource(kTime1, kCsrc2, RtpSourceType::CSRC, 47)));
+
+  constexpr uint32_t kCsrcsSubset[] = {kCsrc1};
+  csrcs.Update(kTime1 + 1, kCsrcsSubset, absl::nullopt);
+  EXPECT_THAT(
+      csrcs.GetSources(kTime1 + 1),
+      UnorderedElementsAre(RtpSource(kTime1 + 1, kCsrc1, RtpSourceType::CSRC),
+                           RtpSource(kTime1, kCsrc2, RtpSourceType::CSRC, 47)));
+}
+
 }  // namespace webrtc
diff --git a/video/rtp_video_stream_receiver.cc b/video/rtp_video_stream_receiver.cc
index 54956688ad..abf88d4371 100644
--- a/video/rtp_video_stream_receiver.cc
+++ b/video/rtp_video_stream_receiver.cc
@@ -320,7 +320,8 @@ void RtpVideoStreamReceiver::OnRtpPacket(const RtpPacketReceived& packet) {
       last_received_rtp_system_time_ms_ = now_ms;
 
       std::vector<uint32_t> csrcs = packet.Csrcs();
-      contributing_sources_.Update(now_ms, csrcs);
+      contributing_sources_.Update(now_ms, csrcs,
+                                   /* audio level */absl::nullopt);
     }
     // Periodically log the RTP header of incoming packets.
     if (now_ms - last_packet_log_ms_ > kPacketLogIntervalMs) {